1. Field of the Invention
The present invention relates to a multiple channel multiplexing apparatus for multiplexing multiple channels of digital audio data in a digital audio interface by applying data compression and expansion technologies.
2. Description of the Prior Art
Increasing the number of audio channels is one trend in modern audio equipment. For example, with the spread of satellite broadcasting in Europe, a single broadcast can cover numerous countries, and demand for multilingual broadcasts has increased as broadcasting areas have expanded. This has strengthened demand for plural audio channel capabilities in commercial video and audio equipment in particular. While two audio channels has been the mainstream in conventional video equipment, four channels are becoming increasingly common, and as market demand grows in the coming years, demand is expected to grow for even more audio channels.
A parallel trend in recent audio equipment is digitization, the most common example of which is the widespread acceptance of the compact disc (CD) format and replacement of LP record libraries with CDs.
Both linear and non-linear quantization are currently used for quantizing analog audio signals. Linear quantization with 16 quantization bits and a sampling frequency of 44.1 kHz is used in the CD format. The digital audio nape (DAT) format uses linear quantization with 16 quantization bits and a sampling frequency of 48 kHz. Products compatible with 20-bit linear quantization are beginning to appear.
A more recent trend in digital audio is the use of digital audio data compression technologies applying digital signal processing technologies. Compression technologies that efficiently compress data by actively using the hearing characteristics of humans to remove unnecessary information, i.e., remove signal components beyond the range of human hearing, have begun to appear in such consumer audio equipment as Digital Compact Cassettes (DCC.sup.(R)) and Mini Disks (MD.sup.(R)). Using the compression/expansion technologies applied in DCC.sup.(R) or MD.sup.(R), the audio transfer rate obtained with the common linear quantization methods used in CDs and bAT can be compressed to 1/4-1/5, and the signal can be expanded during reproduction with virtually no deterioration of audio quality.
For example, while the transfer rate of a two channel audio signal quantized at a sampling frequency of 48 kHz and 16 bits/sample is EQU 48 k.times.16 bits.times.2 channels=1.536 Mbps,
1/4 compression obtains a transfer rate of only 384 Kbps. This is equivalent to the transfer rate of a 4-bit two channel audio signal at a 48-kHz sampling frequency.
Using the digitization technologies and digital audio data compression technologies of modern audio equipment as described above, increasing the number of audio channels should be both possible and practical. One possibility is to use these compression technologies to multiplex a greater number of compressed channels in a digital audio interface used for transferring digital audio data between digital devices. If each channel is quantized at 16 bits and compressed 1/4, and four channels of data are multiplexed together, it is possible to easily increase the number of channels with virtually no change in the transfer format. If this digital audio interface is then connected to a digital recording/reproducing apparatus, all digital recording/reproducing devices with a digital audio interface can be used for plural channel recording and reproducing.
An example of a multiple channel multiplexing apparatus for multiplexing plural channels of compressed digital audio data to a digital audio interface of this type is described below. The digital audio interface multiplexing plural channels in the following example is the digital audio interface defined in Electronic Industries Association of Japan (EIAJ) standard CP-340 (hereafter the AES/EBU digital audio interface), the entire content of which is expressly incorporated by reference herein.
The AES/EBU digital audio interface is described first below.
FIG. 12 is a typical drawing of the signal format of the AES/EBU digital audio interface. Except for part of the channel status information, this signal format is the same for consumer and commercial equipment, and all equipment types and transfer formats. The basic unit is a frame (1 subframe.times.2 channels) with the same repeating frequency as the sampling frequency of the transferred digital audio data, and 192 frames are grouped in one block. Each frame is divided into two subframes containing the data for channels 1 and 2, respectively. Each subframe comprises 32 bits, of which 20 bits are audio data and 4 bits are reserved (AUX) for future bit expansion.
The first 4 bits at the beginning of the subframe are the preamble containing a synchronization signal for indexing and a subframe identification signal. In FIG. 12, B, M, and W indicate the preamble, while M further indicates the first subframe in the frame, W indicates the second subframe in the frame, and B indicates the beginning of the block. The last 4 bits in the subframe are subdata, V being the validity flag indicating whether the transferred data is correct, U being user data, C being the channel status, and P being the parity flag. This channel status forms one word in one block (192 bits), and carries such system information as whether emphasis is applied, the sampling frequency, and whether the transferred data is linearly quantized digital audio data. Part of the channel status information differs in consumer and commercial applications.
It is thus possible co transfer two channels of digital audio data on one line of the AES/EBU digital audio interface. Because the audio data component of each channel is 20 bits, this transfer rate can transfer a maximum 20 bits of data per channel. Therefore, if compression technologies are used and the effective transfer rate after compression is 4 bits, it is possible to multiplex a maximum five channels of data to the audio data component of one channel. If the AUX bits are also used, a maximum of six channels can be multiplexed.
FIG. 13 shows the format of the signal multiplexed to the AES/EBU digital audio interface described above. In this example, four channels of data compressed to 4 bits (shown as channel numbers (1)-(4) in the figure) are muitiplexed to one subframe. Because there are two subframes per frame, a total of 8 channels are multiplexed to one frame.
FIG. 14 shows how the channels are arranged when eight channels are multiplexed to one frame using the multiplexing method shown in FIG. 13. Only the audio data component of each subframe is shown in FIG. 14; the numbers in the blocks indicate the channel number for easier understanding. As shows in FIGS. 13 and 14, all channels are multiplexed to fit in one frame. Because all channels are fit into one frame with this method, there is no difference in the arrangement of data in different frames. Signal processing is therefore simpler than when plural channels are multiplexed across plural frames.
It should be noted, however, that this method is effective when one sample of the 16 (e.g.) quantization bits can be sequentially compressed to the corresponding four bits, i.e., in a compression method in which one sample which is independent of the samples before and after that one sample is also independent after compression. In practice, however, many LSI devices used for compression do not operate in this way. The compressed data is coded data containing coefficients and other information, and four bits do not necessarily represent one sample. As a result, this compressed data does not represent samples of digital audio data. In general, the data compressed in this process is sequentially output from the compression LSI device as single blocks of meaningful data. This output unit is the "data block." In many cases the size of the data block is 16 bits. These data blocks are described in greater detail below.
FIG. 15 is used to describe the data block concept below, and shows the relationship between the linearly quantized input data and the data block output from the compression LSI device. During compression, 16-bit linearly quantized digital audio data is sequentially input to the compression LSI device. While the compression LSI device executes various operations and outputs the operation result, the input/output timing controls operation such that rather than outputting 4 bits for each 16-bit input, 16 bits of compressed data (one data block) containing the compressed data for four samples is output each time four samples of 16-bit linearly quantized data are input. As a result, the time length of the 16-bit data block is four times the time length of the 16-bit linearly quantized data input. This 16-bit data block has meaning only as a single data block, i.e., retains the same least-significant-bit (LSB), most-significant-bit (MSB) concept. The input and output signal flow shown in FIG. 15 is reversed during data expansion, and the data input to the expansion LSI device must be in data block units.
This data block concept is normally used for data compression and expansion.
The problem with this concept is that when a data compression/expansion method having this data block concept in the compressed data is used in a conventional multiple channel multiplexing apparatus as described above, the data block unit is destroyed during the multiplexing operation, and channel multiplexing and demultiplexing cannot be correctly executed.